1. Field of the Invention
The present invention relates to computer networks, and, more particularly, to voice-over-IP networks over wireless local area networks.
2. Description of the Related Art
Traditional telephony carriers, which primarily utilize a public-switched telephony network (“PSTN”), are moving towards a packet-based Voice-over-IP (“VoIP”) infrastructure. A key component of a typical telephony infrastructure is “call control.” Call control comprises a call setup and a call teardown. Both the call setup and the call teardown involve an exchange of call control messages between two end users. Either end user may initiate the setup or teardown. The call setup allocates resources for the exchange of voice and/or data between the two end users. In contrast, the call teardown frees up those resources such that other end users may exchange voice and/or data. In VoIP, call control is achieved through Session Initiation Protocol (“SIP”). It should be noted that one of ordinary skill in the are would contemplate achieving call control through any of a variety of other known protocols.
In addition to carrier networks, VoIP has been steadily ground in enterprise networks as well. In parallel with the adoption of VoIP, many enterprise networks are in the process of deploying support network access via IEEE 802.11 based wireless local area networks (“LANs”). The 802.11 wireless LAN standard offers a medium access method, called Point Coordination Function (“PCF”), that offers support for near-isochronous (i.e., real-time) services where an “Access Point” periodically polls individual stations for packets to transmit. However, there has been little deployment of VoIP over wireless LANs using PCF. A key reason is that most 802.11 Access Points support a medium access method known as Distributed Coordination Function (“DCF”), that is contention-based, i.e., each wireless station competes for control of the wireless medium. While the DCF method works for data packets, VoIP packets, on the other hand, require timely, periodic access to the wireless medium to maintain acceptable voice quality. With increasing use of wireless LANs in the enterprise, use of IP softphones, for example, on 802.11 enabled laptops and handheld devices to initiate and receive VoIP calls will explode.
It is well understood that quality of service (“QoS”) is required for voice traffic in terms of delay, jitter and loss. At the same time, bandwidth on wireless links is far below that of wireline links (e.g., ethernet), and, therefore, uncontrolled access to the wireless medium can introduce unacceptable delay for VoIP traffic.
Therefore, to make efficient use of wireless resources and provide real-time services for VoIP packets, a need exists for a method and apparatus to manage the contention resulting from VoIP call signaling on the wireless medium. Without a solution to this problem, voice quality for VoIP calls over wireless LANs will degrade to unacceptable levels as the data/voice traffic on the wireless link increases. In other words, the method and apparatus should provide sufficient QoS to support wireless voice quality comparable to that of wireline links, even in the prospect of reduced wireless bandwidth.